Hello! I just implemented a jitterbuffer for pjsip in the dialplan in a SBC:
[fromtrunk] exten => _[+0-9]!,1,Set(JITTERBUFFER(fixed)=default) This jitterbuffer catches all calls coming from ISP. My understanding is, that the incoming rtp stream in leg1a is now buffered and handed out "jitter-optimized" to leg2a on the other site (this could be internal or external again). -----------> leg1a leg2a ------------> ISP SBC callee <----------- leg1b leg2b <------------ My question: What's about the rtp stream which is received by leg1b from callee? Is there a receive buffer on the leg1b-site, too? Or is it expected to be done by leg2b before handing it out to leg1b? Iow: is it enough to implement one jitterbuffer? Or should there be a second jitterbuffer on the side of leg2? Thanks for clarification! Regards, Michael -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users